1. Field of the Invention
The present invention relates to automated network traffic management, and more particularly, to prioritization of audio traffic in IP networks. The invention is applicable to both TCP/IP networks, as well as to other IP networks.
2. Description of the Related Art
Currently, there are a number of conventional methods that relate to organization of network traffic management. However all IP-based networks experience packet jams where some packets are dropped or otherwise not delivered. Network channel bandwidth can often be insufficient for passing combined video-audio traffic. Modern video conferencing systems depend on reliable delivery of video-audio packets. The video-audio packets typically jam the channel once the channel capacity changes for a brief moment, and video conference participants lose both picture and sound. However, the sound (i.e., the audio packets) is more important for the purposes of video conferencing. Modern video conferencing systems do not separate audio and video data that is transferred as a combined network media data.
The primary disadvantage of such approach is that when the channel bandwidth is insufficient or becomes temporarily insufficient, the media data requited for video conferencing is not delivered in a consistent manner, and the conference cannot continue. The conventional video conferencing systems do not have means for separating audio and video packets and delivering only the audio data when the bandwidth capacity is low.
Conventional TCP congestion avoidance algorithms determine a current communication channel speed and do not allow a client (or server) to transmit data faster than the channel speed to avoid overload of the network. However, all of the conventional algorithms determine the correct speed only if the communication channel is used to 100% of its capacity. In other cases the current speed is determined incorrectly. This is because two types of congestion avoidance algorithms exist: loss-based (if a packet is lost, then it is assumed that the network is congested) and delay-based (if roundtrip time increases, then it is assumed that the network is congested). If none of these problems are observed, all existing algorithms assume that the network still has additional capacity and increase their estimate of available bandwidth. But on underutilized networks these problems would never be observed, and so the estimate will grow indefinitely.
Accordingly, there is a need in the art for an effective and efficient method for uninterrupted delivery of audio data to video conference participants based on a correct speed of the communication channel.